The present invention relates to an apparatus in which high-frequency band components are newly generated with respect to an input signal, and those are added to a high-frequency band of the input signal to generate a signal with a wideband frequency characteristic.
As is commonly known, a music signal before being transformed into a digital signal continues as spectral envelope so as to naturally attenuate integer-order harmonic overtone series components up to a high-frequency band. Further, there are noninteger-order harmonic components and noise in a music signal. FIGS. 1A and 1B show spectrums before and after an audio signal of certain orchestra music is recorded on a compact disk (hereinafter called CD). As shown in FIG. 1A, it can be understood that high-intensity harmonic components continue up to a high-frequency band at a constant frequency interval in the audio signal before being recorded on the CD. This frequency interval is determined on the basis of a fundamental pitch of a musical instrument.
On the other hand, a frequency band which a digital audio system can reproduce is only half of a sampling frequency to a maximum due to the sampling theorem. For example, as shown in FIG. 1B, in a case of a CD, because its sampling frequency is 44.1 kHz, the frequency characteristic is band-limited to 22.05 kHz.
A characteristic from a sense of listening to a CD sound is expressed by words such as “hard” or “cold” in very many cases as compared to sound from an analog audio system. It has been often pointed out and discussed that such a sense of listening different from that of an analog audio system may be “partly because of band limitations due to the sampling theorem.” Further, a phenomenon called “hypersonic effect,” in which inaudible band sounds greater than or equal to 20 kHz have a favorable effect on the brain's blood flow has been studied and reported. Against this background, there are an increasing number of needs for extending a band of a signal whose high-frequency band is band-limited to a high-frequency band.
As such, proposals and products in which a band is extended by reproducing a lost high-frequency band to improve a sense of listening have already emerged in large numbers. The methods can be roughly divided into “interpolation techniques” and “high-frequency band addition techniques.”
In “interpolation techniques,” after upsampling of an input signal to be doubled or more, a new sample derived by a polynomial interpolation technique or a unique technique is added between samples to extend a frequency band.
The “high-frequency band addition techniques” can be roughly classified into three types as in Patent Document Publications which will be described below.    1. A technique in which an input signal is directly squared and cubed to generate higher harmonics, and only a high-frequency band is taken out of the components thereof, to be added as high-frequency band extension components to the input signal to extend its band (refer to JP2001-356788A).    2. A technique in which components in which an input signal is band-limited by a BPF (Band Pass Filter) is rectified and noise generated by a dither generating circuit are put together to be added as high-frequency band extension components to the input signal (refer to JP-Hei4-245062A).    3. A technique in which an input signal is once transformed into a frequency domain to derive a spectrum, and the derived spectrum is evenly divided into several bands, and a correlation with a band serving as a reference is derived to set a highly-correlated band to high-frequency band extension components, and after those are added to the input signal, this is transformed into a time domain as an output signal (refer to JP2003-15695A).
However, there has been the problem that higher harmonics generated by the “interpolation techniques” are basically folding components of existing components, and are not a reproduction of a correct harmonic overtone series harmonic structure.
On the other hand, there has been the following problems in the “high-frequency band addition techniques.” That is, when the technology disclosed in JP2001-356788A has a harmonic structure composed of only integer-order harmonics due to synthesis of sine waves, it is possible to generate high-frequency band extension components keeping a correct harmonic overtone series interval by square/cube calculations. However, usually, a music signal is hardly composed of only integer-order harmonics, but includes many noninteger-order harmonics and noise components. In particular, in a low-frequency band, there are many noninteger-order harmonic components due to percussion instruments. Therefore, when such a signal is squared, excessive frequency components are generated in large numbers in addition to higher harmonics with a correct harmonic overtone series interval. Further, generally, in an acoustic digital system, because a maximum absolute value amplitude is handled as 1.0 and a minimum absolute value amplitude is handled as 0.0, when a squared calculation is performed, amplitudes of components to be generated exponentially decrease (ex:0.12=0.01). Therefore, when components generated by square/cube calculations onto an input signal are added as high-frequency band extension components to the input signal, there has been a problem that a gap is generated in a spectral envelope, which brings an unnatural frequency characteristic.
In the technology disclosed in JP-Hei4-245062A, in a case in which higher harmonics are generated by nonlinear processing such as full-wave rectification, half-wave rectification, or clipping in a digital system, the higher harmonics generate folding distortion at a Nyquist frequency (the half of a sampling frequency) by necessity. Therefore, there has been a problem that the folding distortion is mixed as unnecessary noise into a band of an original signal.
In the technology disclosed in JP2003-15695A, an FFT (Fast Fourier Transformation) calculation is required in order to make a transformation from a time domain into a frequency domain, and a buffer for frame processing is required. Further, it is necessary to increase the number of FFT points in order to precisely analyze a frequency, and a buffer capacity as well increases with an increase in the number of FFT points. Moreover, an inverse FFT calculation is required in order to transform a signal processed in a frequency domain into a time domain again. As a result, there has been a problem that processing load increases, which may be impossible to mount those with the capability of a DSP mounted in consumer audio equipment.